The invention relates to an apparatus for encoding a wideband digital information signal, the apparatus comprising
an input for receiving the wideband digital information signal, PA1 signal splitting means for, during a specific time interval, splitting the wideband digital information signal into M narrow band sub signals, each one of the M sub signals being representative of a component of the wideband digital information signal which is present in a corresponding one of M adjacent narrow bands in the frequency band of the wideband digital information signal, where M is an integer larger than 1 and the narrow bands all have a specific constant bandwidth, PA1 scale factor determining means for determining a scale factor for subsequent signal blocks in each of the sub signals, PA1 quantization means for quantizing the samples in a signal block into quantized samples in response to bit allocation information supplied to the quantizing means so as to obtain quantized sub signals, PA1 bit allocation information deriving means for deriving said bit allocation information, the bit allocation information being representative of the number of bits with which samples in a signal block of a sub signal will be represented after quantization in the quantization means, PA1 formatting means for combining quantized samples in the signal blocks of the quantized sub signals and scale factors into a digital output signal having a format suitable for transmission or storage, to an apparatus for decoding said coded digital signal so as to obtain a replica of said wideband digital information signal, and to a method for encoding the wideband digital information signal. The wideband digital information signal can be an wideband digital audio signal. PA1 signal block length determining means for determining the lengths of the signal block in at least one of the sub signals and for generating block length information, the block length information being representative of the said lengths of the signal blocks in the at least one sub signal, where the lengths of subsequent signal blocks in said at least one sub signal differ, the scale factor determining means being further adapted to determine the scale factors for subsequent signal block of varying lengths in said at least one sub signal in response to said block length information, the bit allocation information deriving means being further adapted to derive bit allocation information for subsequent signal blocks of varying lengths in said at least one sub signal in response to said block length information, the quantization means being further adapted to quantize the samples in signal blocks of varying lengths in said at least one sub signal in response to said block length information, and the formatting means further being adapted to include the block length information into the digital output signal for transmission or storage. The invention is based on the recognition that the wideband digital information signal may sometimes be of non-stationary character. In that situation, signal transients of short duration are included in the wideband digital signal and are surrounded by signal parts in the wideband digital signal being stationary.
An encoding apparatus as defined in the opening paragraph is known from EP-A 457,390 and EP-A 457,391, to which U.S. Pat. Nos. 5,367,608 and 5,365,553 correspond the documents (D1) and (D2) respectively, in the list of references given below. More specifically, the powers in each of the subbands are calculated by squaring the sample values present in time equivalent signal blocks of the subband signals and summing the squared sample values in a time equivalent signal block. The signal blocks in the documents listed above are of constant length and are 12 samples long.
The powers thus obtained are processed in a processing step in which use is made of a psycho acoustic model so as to obtain masked threshold values. Another way of obtaining the masked threshold values is by carrying out separately a Fourier transform on the wideband digital information signal and applying the psycho acoustic model on the Fourier Transform results. The masked threshold values, together with the scale factor information, result in bitneeds b.sub.1 to b.sub.M for the samples in the time equivalent signal blocks of the M subband signals. In a bitallocation step, those bitneed values are used so as to allocate bits to the samples, resulting in the bitallocation information values n.sub.1 to n.sub.M, n.sub.m indicating the number of bits with which the 12 samples in the signal block of subband m are represented, after having carried out a quantization on the samples in the subbands.
In the prior art encoding system, with a sampling frequency of 48 kHz, the total frequency band to be encoded is 24 kHz. This frequency band is split into 32 narrow bands of equal width, so that they have a constant width of 750 Hz each. The narrow bands may be substantially non-overlapping.
Investigations have resulted in the knowledge that especially in the lower frequency bands, the bandwidth is relatively broad so that, either a large number of bits is required to code the sub signals in that lower frequency bands because of the fact that in some cases the signal-to-mask ratio is large, or, if such large number of bits is not available, encoding errors may become audible upon decoding.
This problem can be solved by decreasing the bandwidth of the subbands, e.g. to half of the original bandwidth, that is to 375 Hz, so that now 64 sub signals will be available at the output of the signal splitting means.